[asterisk-users] Codec preference

[jonas kellens]

Sip.conf :

[general]
;context=default
allowguest=no
allowoverlap=no
allowtransfer=yes
realm=mydomain
bindport=5060
bindaddr=X.X.X.X
maxexpiry=1800 
minexpiry=60 
mohinterpret=default
mohsuggest=default
language=be
useragent=mycorp
dtmfmode = rfc2833 
alwaysauthreject = yes 
;contactdeny=0.0.0.0/0.0.0.0
;contactpermit=172.16.0.0/255.255.0.0
rtptimeout=60
rtpholdtimeout=300
;sipdebug = yes
;recordhistory=yes
;dumphistory=yes 
registertimeout=60
registerattempts=60 
rtcachefriends=yes 
;rtsavesysname=yes  
;rtupdate=yes 
;rtautoclear=yes 
;ignoreregexpire=yes
jbenable = yes
jbforce = no
allowsubscribe=yes
limitonpeer = yes
notifyringing=yes
notifyhold=yes

then come the registrations...


Jonas.


On Fri, 2010-03-12 at 11:47 +0530, Prince Singh wrote:

> Post your Asterisk's sip.conf
> 
> 
> On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens
> <jonas.kellens at telenet.be> wrote:
> 
>         How can I set the prefered codec between 2 calling parties ??
>         
>         My Grandstream supports G729, alaw and gsm... in this order.
>         The Zoiper softphone has alaw and gsm as codecs... in that
>         order.
>         
>         Although there should be a matching codec found, my
>         Grandstream can not call the Zoiper softphone.
>         
>         CLI shows :
>         
>         [Mar 11 17:47:21] WARNING[22367]: channel.c:3340
>         ast_channel_make_compatible: No path to translate from
>         SIP/mygrandstream-09c599e0(2) to SIP/zoiper-09cd57f8(256)
>         [Mar 11 17:47:21]     -- Got SIP response 415 "Unsupported
>         Media Type" back from 192.168.1.106 (<-- zoiper)
>         
>         SIP debug :
>         
>         [Mar 11 17:55:57] Peer audio RTP is at port
>         192.168.1.101:10110 (<-- the Grandstream)
>         [Mar 11 17:55:57] Found audio description format PCMA for ID 8
>         [Mar 11 17:55:57] Found audio description format GSM for ID 3
>         [Mar 11 17:55:57] Found audio description format PCMU for ID 0
>         [Mar 11 17:55:57] Found audio description format G729 for ID
>         18
>         [Mar 11 17:55:57] Found audio description format
>         telephone-event for ID 101
>         [Mar 11 17:55:57] Capabilities: us - 0x10a (gsm|alaw|g729),
>         peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing),
>         combined - 0x10a (gsm|alaw|g729)
>         [Mar 11 17:55:57] Non-codec capabilities (dtmf): us - 0x1
>         (telephone-event), peer - 0x1 (telephone-event), combined -
>         0x1 (telephone-event)
>         ...
>         [Mar 11 17:55:57] Audio is at 192.168.1.150 port 11586 (<-- my
>         Asterisk)
>         [Mar 11 17:55:57] Adding codec 0x100 (g729) to SDP
>         [Mar 11 17:55:57] Adding non-codec 0x1 (telephone-event) to
>         SDP
>         
>         This is what Asterisk sends to the Zoiper in the INVITE
>         (sdp) :
>         
>         Content-Type: application/sdp
>         Content-Length: 263
>         v=0
>         o=root 3208 3208 IN IP4 192.168.1.150
>         s=session
>         c=IN IP4 192.168.1.150
>         t=0 0
>         m=audio 11586 RTP/AVP 18 101
>         a=rtpmap:18 G729/8000
>         
>         Why isn't Asterisk negotiating with the Zoiper for the
>         alaw-codec ??
>         
>         The sip-configuration (realtime MySQL) for the Grandstream
>         is :
>         
>         allow : g729;alaw;gsm
>         
>         and the Zoiper softphone :
>         
>         allow : alaw;gsm;g729
>         
>         
>         Kind regards,
>         
>         Jonas.

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